The present invention relates to communication and recording operations of audio signals, and, more particularly, to an audio signal coding/and decoding method.
In recent years, vigorous development has occurred in the area of high-quantity speech coding for wide band speech used, for example, in videoconferencing or the like, and in the area of high-quality audio coding, used, for example, in multimedia. In such coding methods, an adaptive transform coding method using the spectrum envelope information as side information has frequently been used. This coding/and decoding method using such an adaptive transform coding method is exemplified in the prior art by the method disclosed as "Adaptive Transform Coding Method and System" in Japanese Patent Laid-Open No. 184098/1991, or the method, as disclosed as "Transform Coding of Audio Signals Using Perceptual Noise Criteria" by James D. Johnston: IEEE Journal on Selected Areas in Communications, Vol. 6, No.2".
For preparing the description of the present invention, an adaptive transform coding method which is used in the prior art will first be described. FIG. 2 is a diagram showing a summary of the processing flow of such an adaptive transform coding/and decoding method between a coding transmitter 1 and a decoding receiver 2.
In FIG. 2, reference numeral 3 designates processing of an input buffer for storing a predetermined number of samples of a digitized input, to construct a coding block. Numeral 4 designates processing for transforming the input audio signal to a frequency domain by a fast Fourier transform or the like providing an output of a plurality of discrete frequency bands. Numeral 13 designates processing for inverse transforming to a time domain, as corresponds to the transformation 4. Numeral 5 designates processing for quantizing the transform coefficient by using a Max. quantizer, and numeral 11 designates processing for an inverse quantization, as corresponds to the quantization 5. Numeral 6 designates processing for calculating a spectrum envelope. This can be done, for example, by using a method of approximating the spectrum envelope by averaging the powers of the transform coefficients of the frequency domain for several discrete frequency bands, a method of deducing the spectrum envelope by linearly predictively analyzing the input, and so on. Numeral 7 designates processing for coding the spectrum envelope, and numeral 12 designates processing for decoding the spectrum envelope, as corresponds to the coding 7. Numeral 8 designates processing for adaptively controlling the bit allocation/and quantization step size of the transform coefficient quantization of each discrete frequency band on the basis of the rate-distortion theorem or the like. Numeral 9 designates processing for multiplexing the quantization transform coefficient and the spectrum envelope code to generate a transmission code, and numeral 10 designates processing for demultiplexing the transmission code to decode the quantization transform coefficient and the spectrum envelope code. Numeral 14 designates an output buffer for storing the output signals as the unit of a block to output them sequentially.
The coding/and decoding flow will be described with reference to FIG. 2. In the coding operation, a coding block is constructed from the input audio signal by the buffer 3 and is transformed into a transform coefficient by the frequency domain transformation 4 until it is quantized by the transform coefficient quantization 5. In this transform coefficient quantization 5, the coefficient of each discrete frequency band is quantized with a bit allocation and a quantization step size which are adaptively controlled on the basis of the spectrum envelope obtained by the spectrum envelope calculation 6 from the input signal. These operations are accomplished for auditorily controlling the quantization distortion of each discrete frequency band. On the other hand, the full band of the spectrum envelope is coded by the coding operation 7. Then, the transmission code is generated from the quantization transform coefficient and the spectrum envelope code by the multiplexing operation 9.
In the decoding operation 2, the quantization transform coefficient and the spectrum envelope code are separated at first by the demultiplexing operation 10. Then, the spectrum envelope is decoded by the spectrum envelope decoding operation 12, and the bit allocation/and quantization step size are calculated by the bit allocation/and quantization step size calculation 8 on the basis of the decoded spectrum envelope so that the transform coefficient is decoded in the inverse transform coefficient quantization 11 by applying the bit allocation/and quantization step size. This coefficient is transformed into a time-domain signal by the inverse time domain transformation 13 and is stored in the output buffer 14 so that it is sequentially outputted to decode the audio signal.
In the adaptive transform coding method described above as prior art, the full band of the spectrum envelope is coded by an identical coding method and is updated for each full-band block. On the other hand, the time fluctuation of the spectrum envelope of the audio signal can be different for different bands within the full band, and generally has a tendency that the time fluctuation diminishes in the lower frequency domain. Thus, a band with a small time fluctuation has a large correlation between the adjoining blocks and a large redundancy. However, this redundancy is not effectively exploited to have a low coding efficiency by the adaptive transform coding method of the prior art, in which the spectrum envelope is coded by the identical coding method for the full band and is updated at each full-band block. Especially in case the spectrum envelope is to be estimated by linear predictive analysis, the method of the prior art has been unable to consider the differences of the time fluctuation for each band, because the input signal is analyzed as a whole for the full band so that the linear prediction coefficient is calculated/and coded and transmitted.
In the prior art, as described above, there is no consideration of redundancy which is caused by the difference in the time fluctuation of the spectrum envelope between the bands. Hence, the prior art is insufficient for providing an adaptive transform coding/and decoding method for low bit-rate coding of high quality.